In a digital telephone system, digital speech signal is processed in speech coding e.g. by frames of about 20 ms and using various methods, so that the result is a set of speech-depicting parameters per each frame. This information or the set of parameters is channel coded and sent to the transmission path.
FIG. 1 is a simplified view of a GSM network from the viewpoint of transmission. A Network Subsystem NSS includes a Mobile Switching Centre 15 MSC, through the system interface of which the mobile telephone network is connected to other networks, such as a Public Switched Telephone Network PSTN. The Network Interworking Functions 16 IWF are designed to adapt the GSM network to these other networks. The network subsystem NSS is joined through an A interface with a Base Station Subsystem BSS, which includes Base Station Controllers BSC 14, each of which controls the Base Transceiver Stations BTS 13 connected to them. The interface between the base station controller and the base transceiver stations connected to it is an A bis interface. The base transceiver stations again are in radio communication with the mobile stations through a radio interface.
Speech coding is done on both sides of the radio interface, that is, in the mobile station and in the network. The speech encoder located on the network side is called a transcoder and it is a part of a Transcoder and Rate Adaptation Unit TRAU. This unit is part of a base station subsystem BSS and it may be located in connection with the base station controller 14 as in FIG. 1 or in connection with the mobile services switching centre. The transcoders convert the speech from digital format into some other format, e.g. they convert 64 kbit/s A-law PCM coming over the A interface from the centre into 13 kbit/s Full Rate FR coded speech for conveyance to the base station line and vice versa. If the transmission is data transmission, they also perform rate adaptation.
On the network side, the transcoder unit must obtain information from the radio interface for efficient decoding. For this purpose a special inband signaling is used on that same channel between the base transceiver station and the transcoder unit, on which speech or data is transmitted.
The FR speech coder input is either a 13 bit PCM signal arriving from the network side or A/D converted 13 bit PCM arriving from the audio part of the mobile station. The speech frame, a so-called TRAU frame, obtained from the coder output, has a duration of 20 ms and it includes 260 bits, which are formed by coding 160 PCM-coded speech samples. In addition, the TRAU frame has 60 bits available for frame synchronisation, for indication of speech and data, for timing and for other information, so that the total length of the TRAU frame is 320 bits. The frame produced by the Half Rate HF speech coder includes 112 bits, which is equal to a bit flow of 5,6 kbit/s. The frame produced by an Enhanced Full Rate EFR includes 244 bits, which is equal to a bit flow of 12,2 kbit/s.
FIG. 1 shows the transmission rates per channel which are used in the GSM. The mobile station sends speech or data information over the radio interface on a radio channel as traffic frames. Base transceiver station 13 receives the information and transmits it to the TRAU frame into the subtime slot of the PCM line. The full rate connection uses a 16 kbit/s TRAU frame while the half rate connection uses an 8 kbit/s TRAU frame. In the same 64 kbit/s time slot of the PCM line, 4 or 8 traffic channels are obtained in this way. In base station controller 14 the transcoder/rate adaptation unit TRAU converts the digital information contained in the TRAU frame to a rate of 64 kbit/s, and the data is transmitted at this rate to the mobile services switching centre, whereupon following modulation and a rate change, if such are required, the information is transmitted to some other network.
In the arrangement described above, a double speech coding in a call between two mobile stations takes place in the network: the speech coming from the first mobile station which is transmitted in TRAU frames and coded in the mobile station's speech coder is decoded in the TRAU unit, whereupon the speech propagates as PCM speech to the mobile services switching centre and thence further e.g. to a TRAU unit which is located in another location area of the same centre area and which codes the speech into TRAU frames. This double coding is called tandem coding. If the mobile stations use the same speech codec, any tandem coding between them is unnecessary as such and will only result in a poorer quality of speech. Various methods of preventing tandem coding have therefore been proposed. It is a common feature of most of these that tandem coding is prevented by sending the coded speech parameters in the PCM time slot on a sub-channel formed by one or several less significant bits while the remaining bits are used for sending the most significant bits of the PCM samples. If the TRAU unit at the other end perceives that there are TRAU frames on the sub-channel, it will not perform coding on them, but it will transmit them further as such, so the parameters will be decoded to speech only in the mobile station. If the transcoder is unable to identify the sub-channel, it will code the PCM samples in a normal manner into speech parameters. The methods differ from each other mainly in how the transcoders know not to encode samples transmitted on the sub-channel. The information can be relayed in hand shaking between the transcoders, by monitoring the sub-channel's signalling pattern or by having the mobile services switching centre inform the transcoders about the matter.
Current mobile telephone networks allow speech transmission between two parties, each of which may be a subscriber of a mobile telephone network as the call travels inside the same mobile telephone network or by way of a circuit switched PSTN/ISDN network from one mobile telephone network to another. The other subscriber may also be a subscriber of a circuit switched PSTN/ISDN network. In all cases, the connection is always circuit switched and it is reserved for use by these two parties during the whole transmission of information.
Originally, the mobile telephone network was designed for efficient speech transmission, and in present day networks data transmission rates are in fact rather low. This situation is improved by the General Packet Radio Service GPRS system which uses virtual circuits and is intended for transmission of packet data and which is being specified by the ETSI (European Telecommunications Standards Institute). The purpose of GPRS services is to operate independently of current circuit switched services, and especially to utilize the unused resources of circuit switched traffic.
Based on the foregoing it is possible to perceive the matter which is a characteristic of current mobile telephone networks, that speech calls going out from a mobile services switching centre always travel to a public switched telephone network PSTN.
Lately, use of the Internet for speech transmission has become increasingly popular. In telephone traffic transmitting speech in packet form over a data network the terminal equipment used is e.g. a computer which is equipped with multimedia properties and is connected to a telecommunication network. The connection with another party is set up by way of the Internet network. Call management operations, such as call set up and release, are performed with the aid of software in the computer. The call is controlled by a CTI programme (CTI=Computer Telephony Integration), which reads the speech going out from the audio card and converts it into a form suitable for conveyance in a telecommunication network. The programme reads the incoming speech from a signal received from the telecommunication network and controls the audio card to repeat the speech from loudspeakers. The programme also maintains information on the state of the call, such as whether the terminal equipment is in a passive (on-hook) or in an active (off-hook) call state, whether the call is going on, whether a call is coming in, whether dialling is being done, etc. and through the telecommunication network it forms a connection with the other participant or participants to the call. An advantage of the call is its cheap price, but it suffers especially from the poor call quality caused by transmission delays in particular.
The separateness of networks is a problem with calls of the Internet and the mobile telephone network. By using the speech service of a mobile station network, the mobile station subscriber can not get in speech connection with a piece of terminal equipment connected to the Internet packet network, nor can the terminal equipment have a connection with the mobile station.
The present invention aims at a system, which allows the described speech calls to a data network, especially between a telephone connected to the Internet and a mobile station.
The proposed system is characterised in that which is stated in the independent claims.